[Asterisk-Users] *8 problem still there?
Luis Vazquez
luis at teledata.com.uy
Wed May 19 08:47:31 MST 2004
Shaun Ewing wrote:
>I'm not seeing this - using stable CVS from 14-05-2004.
>
>Phone types are Grandstream Budgetone 101 using firmware 1.0.4.68, and Cisco
>7940 using SIP 6.2.
>
>-Shaun
>
>
>
Just to give more info.
I just made a testing using stable CVS from 24-04-2004 and 3 softphone
clients registered in asterisk with users
luis, lia and jorge (with fromdomain=ipcontact.com.uy in sip.conf):
kphone ( sip:111 ---> sip:luis at 192.168.2.175 ---> sip:192.168.2.176:5062)
messenger ( sip:114 --> sip:lia at 192.168.2.175 ---> sip:192.168.2.179:16616 )
xlite ( sip:jorge at 192.168.2.175 ---> sip:192.168.2.179:5061)
Here is the dialog in a call from luis(kphone) to 114(messenger) and a
pickup with *8 from jorge(xlite).
The kphone and xlite get connected but 114 (lia - messenger) never gets
a CANCEL:
****Invite from luis to 114 at asterisk **************:
U 192.168.2.176:5062 -> 192.168.2.175:5060
INVITE sip:114 at 192.168.2.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.176:5062;rport
CSeq: 5406 INVITE
To: <sip:114 at 192.168.2.175>..Content-Type: application/sdp
From: "Luis Vazquez" <sip:luis at 192.168.2.175>;tag=E340D0A
Call-ID: 1687598931 at 192.168.2.176
Subject: sip:luis at 192.168.2.175
Content-Length: 187
User-Agent: kphone/4.0.2
Contact: "Luis Vazquez" <sip:luis at 192.168.2.176:5062;transport=udp>
v=0..o=username 0 0 IN IP4 192.168.2.176..s=The Funky Flow
c=IN IP4 192.168.2.176..t=0 0
m=audio 32842 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000..
#
U 192.168.2.175:5060 -> 192.168.2.176:5062
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.176:5062;rport;received=192.168.2.176
From: "Luis Vazquez" <sip:luis at 192.168.2.175>;tag=E340D0A
To: <sip:114 at 192.168.2.175>;tag=as38ce4ffc
Call-ID: 1687598931 at 192.168.2.176
CSeq: 5406 INVITE
User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:114 at 192.168.2.175>
Content-Length: 0....
#
********** Relay of Invite from asterisk to messenger***************:
U 192.168.2.175:5060 -> 192.168.2.179:16616
INVITE sip:192.168.2.179:16616 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.175:5060;branch=z9hG4bK535d9d2c
From: "Luis(1084976431.475)" <sip:111 at ipcontact.com.uy>;tag=as3d3529c2
To: <sip:192.168.2.179:16616>
Contact: <sip:111 at 192.168.2.175>
Call-ID: 4b188f24540489523fc9751f124e8068 at 192.168.2.175
CSeq: 102 INVITE
User-Agent: Asterisk PBX..Date: Wed, 19 May 2004 14:20:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
UniqueID: 1084976433.476
Content-Type: application/sdp
Content-Length: 211
v=0
o=root 20766 20766 IN IP4 192.168.2.175
s=session
c=IN IP4 192.168.2.175..t=0 0
m=audio 17996 RTP/AVP 0 397
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -..
#
*********** Asterisk says to kphone messenger is ringing **************:
U 192.168.2.175:5060 -> 192.168.2.176:5062
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.176:5062;rport;received=192.168.2.176
From: "Luis Vazquez" <sip:luis at 192.168.2.175>;tag=E340D0A
To:<sip:114 at 192.168.2.175>;tag=as38ce4ffc..Call-ID: 1687598931 at 192.168.2.176
CSeq: 5406 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:114 at 192.168.2.175>
Content-Length: 0
#
********** Messenger says to Asterisk he is trying ******************:
U 192.168.2.179:1071 -> 192.168.2.175:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.175:5060;branch=z9hG4bK535d9d2c
From: "Luis(1084976431.475)" <sip:111 at ipcontact.com.uy>;tag=as3d3529c2
To: <sip:192.168.2.179:16616>;tag=b271370b-aeed-4640-adca-d60c86b188d7
Call-ID: 4b188f24540489523fc9751f124e8068 at 192.168.2.175
CSeq: 102 INVITE
User-Agent: Windows RTC/1.0
Content-Length: 0
#
*********** Messenger is ringing (and will be forever if not anwered)
****************:
U 192.168.2.179:1071 -> 192.168.2.175:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.175:5060;branch=z9hG4bK535d9d2c
From: "Luis(1084976431.475)" <sip:111 at ipcontact.com.uy>;tag=as3d3529c2
To: <sip:192.168.2.179:16616>;tag=b271370b-aeed-4640-adca-d60c86b188d7
Call-ID: 4b188f24540489523fc9751f124e8068 at 192.168.2.175
CSeq: 102 INVITE
User-Agent: Windows RTC/1.0
Content-Length: 0
#
********* Here starts call pickup ***************
*********** Xlite enters the game sending an Invite to *8 at asterisk *******:
U 192.168.2.179:5061 -> 192.168.2.175:5060
INVITE sip:*8 at ipcontact.com.uy SIP/2.0
Via: SIP/2.0/UDP
192.168.2.179:5061;rport;branch=z9hG4bK41344F5B08434749A915F0DCFAB2AE66
From: Jorge <sip:jorge at ipcontact.com.uy:5061>;tag=1940958518
To: <sip:*8 at ipcontact.com.uy>
Contact: <sip:jorge at 192.168.2.179:5061>
Call-ID:BE331F89-45C6-48DA-85F8-F3AD91A827A7 at 192.168.2.179
CSeq: 19484 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103a
Content-Length: 193
v=0..o=jorge 2391140 2391203 IN IP4 192.168.2.179
s=X-Lite
c=IN IP4 192.168.2.179..t=0 0
m=audio 8000 RTP/AVP 3 101
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15..
#
******** Asterisk responds he is trying **************:
U 192.168.2.175:5060 -> 192.168.2.179:5061
SIP/2.0 100 Trying
Via:
SIP/2.0/UDP192.168.2.179:5061;rport;branch=z9hG4bK41344F5B08434749A915F0DCFAB2AE66
From: Jorge <sip:jorge at ipcontact.com.uy:5061>;tag=1940958518
To: <sip:*8 at ipcontact.com.uy>;tag=as4b041d55
Call-ID: BE331F89-45C6-48DA-85F8-F3AD91A827A7 at 192.168.2.179
CSeq: 19484 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*8 at 192.168.2.175>
Content-Length: 0
#
********* Asterisk accept the call from the Xlite (jorge) *****:
U 192.168.2.175:5060 -> 192.168.2.179:5061
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.179:5061;rport;branch=z9hG4bK41344F5B08434749A915F0DCFAB2AE66
From: Jorge <sip:jorge at ipcontact.com.uy:5061>;tag=1940958518
To: <sip:*8 at ipcontact.com.uy>;tag=as4b041d55
Call-ID:BE331F89-45C6-48DA-85F8-F3AD91A827A7 at 192.168.2.179
CSeq: 19484 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*8 at 192.168.2.175>
UniqueID:1084976436.477
Content-Type: application/sdp
Content-Length: 215
v=0
o=root 3611 3611 IN IP4 192.168.2.175
s=session
c=IN IP4 192.168.2.175
t=0 0
m=audio 16578 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -..
#
****** Asterisk accept the call from kphone and bridges with xlite ******:
U 192.168.2.175:5060 -> 192.168.2.176:5062
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.176:5062;rport;received=192.168.2.176
From: "Luis Vazquez"<sip:luis at 192.168.2.175>;tag=E340D0A
To: <sip:114 at 192.168.2.175>;tag=as38ce4ffc
Call-ID: 1687598931 at 192.168.2.176
CSeq: 5406 INVITE..User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:114 at 192.168.2.175>..UniqueID: 1084976431.475
Content-Type: application/sdp
Content-Length: 162
v=0
o=root 20766 20766 IN IP4 192.168.2.175
s=session
c=IN IP4 192.168.2.175
t=0 0
m=audio 16964 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off
#
***************** Both clients sed theirs ACKs and get connected
**************:
U 192.168.2.176:5062 -> 192.168.2.175:5060
ACK sip:114 at 192.168.2.175 SIP/2.0..Via: SIP/2.0/UDP
192.168.2.176:5062;rport
CSeq: 5406 ACK
To: <sip:114 at 192.168.2.175>;tag=as38ce4ffc
From: "Luis Vazquez" <sip:luis at 192.168.2.175>;tag=E340D0A
Call-ID:1687598931 at 192.168.2.176
Content-Length: 0..User-Agent: kphone/4.0.2
Contact: "Luis Vazquez" <sip:luis at 192.168.2.176:5062;transport=udp>
#
U 192.168.2.179:5061 -> 192.168.2.175:5060
ACK sip:*8 at 192.168.2.175 SIP/2.0..Via: SIP/2.0/UDP
192.168.2.179:5061;rport;branch=z9hG4bK4CE63D00AB944D4CB7BED0D3A2B8B939
From: Jorge <sip:jorge at ipcontact.com.uy:5061>;tag=1940958518
To: <sip:*8 at ipcontact.com.uy>;tag=as4b041d55
Contact: <sip:jorge at 192.168.2.179:5061>
Call-ID: BE331F89-45C6-48DA-85F8-F3AD91A827A7 at 192.168.2.179
CSeq: 19484 ACK..Max-Forwards: 70
Content-Length: 0
What happened with our friend lia at messenger????
She is still ringing and waiting for a CANCEL a BYE or something.
And that's all.
Just in case here is the sip.conf
[general]
port = 5060
context = local
..........
[luis]
type = friend
callgroup=2
pickupgroup=2
username = luis
host = dynamic
disallow=all
allow=ulaw
allow=gsm
dtmfmode=inband
callerid="Luis" <111>
[jorge]
type = friend
callgroup=2
pickupgroup=2
username = jorge
disallow=all
allow=gsm
dtmfmode=rfc2833
host = dynamic
callerid="Jorge" <112>
[lia]
type = friend
callgroup=2
pickupgroup=2
username = lia
dtmfmode=inband
host = dynamic
callerid="Lia" <114>
I hope someone have the time and patience to take a look.
Godbye
Luis
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